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gerrit-public.fairphone.software / platform / external / webrtc / 27c293665e3a2b0e2de12236529773a7b9057f6e / media / engine
  1. fcf79cc Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats. by Åsa Persson · 4 years, 9 months ago
  2. 03fbace Remove apm_helpers, consolidate audio config in WebRtcVoiceEngine by Sam Zackrisson · 4 years, 9 months ago
  3. 9429888 Delete deprecated bytes_sent/bytes_rcvd stat values by Niels Möller · 4 years, 10 months ago
  4. 0bad15f Remove the noise_suppression() pointer to submodule interface by saza · 4 years, 10 months ago
  5. 8038541 Update the header extensions capabilities with mid, rid and rrid by Florent Castelli · 4 years, 10 months ago
  6. ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 4 years, 10 months ago
  7. 35214fc Add missing RTC_EXPORT for the component build. by Mirko Bonadei · 4 years, 10 months ago
  8. ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 4 years, 10 months ago
  9. fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 4 years, 10 months ago
  10. 80f53b7 Extend WebRTC-Video-MinVideoBitrate to experiment per-codec by Elad Alon · 4 years, 10 months ago
  11. ff27da5 Add/remove receive streams with SSRC 0 from media channels by Saurav Das · 4 years, 10 months ago
  12. f4e0c29 SimulcastEncoderAdapter: support per layer fallback and single encoder proxying by Erik Språng · 4 years, 10 months ago
  13. 9d7eb28 Don't limit simulcast layers number for screenshare based on resolution by Ilya Nikolaevskiy · 4 years, 10 months ago
  14. 09f1195 Always pass arguments to INSTANTIATE_TEST_SUITE_P. by Mirko Bonadei · 4 years, 10 months ago
  15. 27b0e0d Remove obsolete todo comment in simulcast.h by Åsa Persson · 4 years, 10 months ago
  16. 1b83a9e Only handle each RTCP once. by Sebastian Jansson · 4 years, 10 months ago
  17. 53227cc Remove webrtc::MinPositive from api/. by Mirko Bonadei · 4 years, 10 months ago
  18. 738bfa7 Remove api/bitrate_constraints.h. by Mirko Bonadei · 4 years, 10 months ago
  19. 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 4 years, 10 months ago
  20. d9cc8c0 Encoder switching based on network and/or resolution conditions. by philipel · 4 years, 10 months ago
  21. 73ceed5 Update simulcast bitrate calculations for non-standard resolutions. by Ilya Nikolaevskiy · 4 years, 10 months ago
  22. 7bf7a42 Delete flag VideoReceiveStream::Config::Rtp::remb by Niels Möller · 4 years, 11 months ago
  23. 65f17ca Move MediaTransportInterface out of the libjingle_peerconnection_api target by Niels Möller · 4 years, 11 months ago
  24. cc62b16 Add qualityLimitationResolutionChanges stat by Evan Shrubsole · 4 years, 11 months ago
  25. 8c5520c Reland "Make the min video bitrate in VideoSendStream configurable." by Ying Wang · 5 years ago
  26. 1d2149c Revert "Make the min video bitrate in VideoSendStream configurable." by Alessio Bazzica · 5 years ago
  27. b2fb0b9 Make the min video bitrate in VideoSendStream configurable. by Ying Wang · 5 years ago
  28. a837030 Split out RtpSource from libjingle_peerconnection_api by Niels Möller · 5 years ago
  29. 25eb47c Make the RtpHeaderParserImpl available to tests and tools only. by Tommi · 5 years ago
  30. 4271afb Fix the bug and reland "Make min video target bitrate configurable." by Ying Wang · 5 years ago
  31. 0c141c5 Fix frames dropped statistics by Johannes Kron · 5 years ago
  32. 7e896d0 Revert "Make min video target bitrate configurable." by Mirko Bonadei · 5 years ago
  33. a471e79 Make min video target bitrate configurable. by Ying Wang · 5 years ago
  34. d77cc24 New const method StreamStatistician::GetStats by Niels Möller · 5 years ago
  35. 224c69d Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo by Niels Möller · 5 years ago
  36. 70efdde Set local ssrc at construction of Rtp module by Erik Språng · 5 years ago
  37. bbeb109 Reporting audio device underrun counter by Alex Narest · 5 years ago
  38. 71c6b56 Allow sending abs-send-time for audio streams. by Sebastian Jansson · 5 years ago
  39. 78a7138 Remove MediaTransport from Call. by Tommi · 5 years ago
  40. 5b5d97c Reland of "Reporting of decoding_codec_plc events"" by Alex Narest · 5 years ago
  41. 2d2bbb1 Filter out duplicate receive codecs in the media engine by Steve Anton · 5 years ago
  42. 83bbe91 Delete deprecated rtc_event_log header by Danil Chapovalov · 5 years ago
  43. f40a340 Remove deprecated code related to AEC2 by Per Åhgren · 5 years ago
  44. d2845f8 Removes unused AudioAllocationSettings from voice engine. by Sebastian Jansson · 5 years ago
  45. 9b1700c Enable field trial LegacySimulcastLayerLimit by default by Florent Castelli · 5 years ago
  46. d7ee76c Wire up field trials for some experimental screenshare settings by Erik Språng · 5 years ago
  47. 8bbdb5b Update VideoBitrateAllocator allocate to take a struct with more fields by Florent Castelli · 5 years ago
  48. da4f093 Reland "Only include payload in bytes sent/received." by Bjorn A Mellem · 5 years ago
  49. bedb7a8 Revert "Reporting of decoding_codec_plc events" by Mirko Bonadei · 5 years ago
  50. bcd068d Revert "Only include payload in bytes sent/received." by Bjorn Mellem · 5 years ago
  51. 0a88ea0 Reporting of decoding_codec_plc events by Alex Narest · 5 years ago
  52. a9fbb22 Add a field trial for older applications to reduce the simulcast layer count by Florent Castelli · 5 years ago
  53. 74a1b4b Only include payload in bytes sent/received. by Bjorn A Mellem · 5 years ago
  54. 0182a03 Reland "Remove the injectable bitrate allocation strategy API." by Jonas Olsson · 5 years ago
  55. e95b57c Revert "Remove the injectable bitrate allocation strategy API." by Mirko Bonadei · 5 years ago
  56. 0bb0881 Add VideoEncoderFactory::GetImplementations function. by philipel · 5 years ago
  57. 66b3860 Remove WebRTC-SimulcastScreenshare and enable it by default by Florent Castelli · 5 years ago
  58. 80cb3f6 Remove the injectable bitrate allocation strategy API. by Jonas Olsson · 5 years ago
  59. 495a1ae Remove cricket::WebRtcMediaEngineFactory as now unused by Danil Chapovalov · 5 years ago
  60. a4d8737 Format almost everything. by Jonas Olsson · 5 years ago
  61. 668ce0c Remove trial WebRTC-SimulcastMaxLayers and make its behavior default by Florent Castelli · 5 years ago
  62. fdf74bd Remove non implemented function from WebRtcVideoChannel. by philipel · 5 years ago
  63. 53d45ba Make TaskQueueFactory required construction parameter for Call by Danil Chapovalov · 5 years ago
  64. e8ed830 WebRtcVideoChannel encoder fallback. by philipel · 5 years ago
  65. 5ee6967 Don't reset encoder on max/min bitrate change. by Sergey Silkin · 5 years ago
  66. bfd343b Add totalDecodeTime to RTCInboundRTPStreamStats by Johannes Kron · 5 years ago
  67. 5983087 Forced vp8 sw encoder fallback: only use min bitrate config if codec type is vp8. by Åsa Persson · 5 years ago
  68. 65764e4 Add missing overrides in VideoEncoder proxies/adapters by Elad Alon · 5 years ago
  69. 8f01c4e Define FecControllerOverride and plumb it down to VideoEncoder by Elad Alon · 5 years ago
  70. 2efae77 Add RTCStats for keyFramesEncoded, keyFramesDecoded. by Rasmus Brandt · 5 years ago
  71. 90f3b89 Replace the implementation of `GetContributingSources()` on the video side. by Chen Xing · 5 years ago
  72. 3472b9a Delete RTCInboundRTPStreamStats::fraction_lost by Niels Möller · 5 years ago
  73. c538506 Enable H.264 temporal scalability in simulcast. by Johnny Lee · 5 years ago
  74. 98cbb22 Moved AsyncInvoker to be destructed first in WebRtcVideoSendStream. by philipel · 5 years ago
  75. a9952cb Uncomment "override" in simulcast_encoder_adapter_unittest.cc by Elad Alon · 5 years ago
  76. 370f93a Reland "Inform VideoEncoder of negotiated capabilities" by Elad Alon · 5 years ago
  77. e8e4dc4 Change StartAecDump methods to work with FILE* and FileWrapper by Niels Möller · 5 years ago
  78. 49d661a Revert "Inform VideoEncoder of negotiated capabilities" by Philip Eliasson · 5 years ago
  79. 11dfff0 Inform VideoEncoder of negotiated capabilities by Elad Alon · 5 years ago
  80. 4fc0855 Cleanup video frame metadata copying by Ilya Nikolaevskiy · 5 years ago
  81. 479a3c0 Add support for enabling and negotiating raw RTP packetization. by Mirta Dvornicic · 5 years ago
  82. 2dbc627 Check H264 packetization mode when using IsSameCodec by Steve Anton · 5 years ago
  83. 85b8ce2 In media engine replace forward declaration with proper includes by Danil Chapovalov · 5 years ago
  84. 220f4be Remove some media/ --> pc/ test dependencies by Steve Anton · 5 years ago
  85. 6737841 Add jitterBufferDelay and jitterBufferEmittedCount stats for video by Guido Urdaneta · 5 years ago
  86. ce33b6a Implement QualityLimitationReasonTracker and expose "reason". by Henrik Boström · 5 years ago
  87. 6e436d1 [audio] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo by Henrik Boström · 5 years ago
  88. 87e3f9d [video] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo by Henrik Boström · 5 years ago
  89. fadb181 Negotiate use of RTCP loss notification feedback (LNTF) by Elad Alon · 5 years ago
  90. 316f3ac Datagram Transport Integration by Anton Sukhanov · 5 years ago
  91. 040dc43 Fix shadowing of override_field_trials_ in WebRtcVideoEngineTest by Elad Alon · 5 years ago
  92. 23aff9b Implement RTCOutboundRtpStreamStats.totalEncodedBytesTarget. by Henrik Boström · 5 years ago
  93. 4f08faa Introduce MediaTransportConfig by Anton Sukhanov · 5 years ago
  94. 19da5ce Formatting of WebRTC-Vp9InterLayerPred field trial. by Sergey Silkin · 5 years ago
  95. 9fe1834 Implement RTCOutboundRtpStreamStats.totalPacketSendDelay for video. by Henrik Boström · 5 years ago
  96. fe4f694 Add missing overrides to QualityTestVideoEncoder by Elad Alon · 5 years ago
  97. 9356252 Ensure that we always set values for min and max audio bitrate. by Daniel Lee · 5 years ago
  98. 8d8ffdb Expose new audio stats on the API by Ivo Creusen · 5 years ago
  99. 44125fa Reland "Piping audio interruption metrics to API layer" by Henrik Lundin · 5 years ago
  100. fc02a79 Revert "Piping audio interruption metrics to API layer" by Henrik Andreassson · 5 years ago